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gst-plugins-base 1.6.1

ChangeLog
=========

2015-10-30  Sebastian Dröge <slomo coaxion net>

        * configure.ac:
          releasing 1.6.1

2015-10-30 14:27:21 +0200  Sebastian Dröge <sebastian centricular com>

        * po/nl.po:
        * po/zh_CN.po:
          po: Update translations

2015-10-28 11:32:57 +0100  Wim Taymans <wtaymans redhat com>

        * gst/audioconvert/gstaudioquantize.c:
          audioquantize: fix 8-pole noise shaping
          Fix the 8-pole noise shaping error update. We were mixing errors from
          different channels.

2015-10-27 15:44:06 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/playback/gstdecodebin2.c:
          decodebin: Send SEEK events directly to adaptive streaming demuxers
          This makes sure that they will always get SEEK events, even if we're currently
          in the middle of a group switch (i.e. switching to another
          representation/bitrate/etc).
          https://bugzilla.gnome.org/show_bug.cgi?id=606382

2015-10-06 15:20:51 +0200  Guillaume Desmottes <guillaume desmottes collabora co uk>

        * gst/playback/gstdecodebin2.c:
          decodebin: fix event leak
          As stated in GST_PAD_PROBE_HANDLED's documentation, we are
          supposed to unref the event before returning.
          Fixes an event leak in the validate.hls.playback.play_15s.hls_bibbop
          validate scenario.
          https://bugzilla.gnome.org/show_bug.cgi?id=754459

2015-10-23 12:02:28 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/playback/gstplaybin2.c:
          playbin: Send upstream events directly to playsink
          Send event directly to playsink instead of letting GstBin iterate
          over all sink elements. The latter might send the event multiple times
          in case the SEEK causes a reconfiguration of the pipeline, as can easily
          happen with adaptive streaming demuxers.
          What would then happen is that the iterator would be reset, we send the
          event again, and on the second time it will fail in the majority of cases
          because the pipeline is still being reconfigured

2015-10-22 12:07:44 +0800  Pavel Bludov <pbludov gmail com>

        * gst-libs/gst/video/video-overlay-composition.h:
          video: overlay-composition: fix rectangle and composition cast macros
          Closing parenthesis was missing in two cases.
          https://bugzilla.gnome.org/show_bug.cgi?id=756893

2015-10-20 12:08:23 +0300  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/rtp/gstrtpbuffer.h:
          rtp: GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING is Since 1.6.1

2015-10-16 10:48:50 +1100  Matthew Waters <matthew centricular com>

        * win32/common/libgstrtp.def:
          win32 update exports for new rtp symbols

2015-07-22 11:31:05 +0200  Stian Selnes <stian pexip com>

        * gst-libs/gst/rtp/gstrtpbuffer.c:
        * gst-libs/gst/rtp/gstrtpbuffer.h:
        * tests/check/libs/rtp.c:
          rtpbuffer: Add map flag to skip padding
          Encrypted RTP buffers may contain encrypted padding, hence it's
          necessary to have an option to relax the validation in order to
          successfully map the buffer.
          When the flag GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING is set
          gst_rtp_buffer_map() will map the buffer like if padding is not
          present.
          https://bugzilla.gnome.org/show_bug.cgi?id=752705

2015-09-14 15:25:11 +0900  eunhae choi <eunhae1 choi samsung com>

        * gst-libs/gst/audio/gstaudiobasesink.c:
          audiobasesink: fix issue about eos handling during flushing
          If the flush-start is arrived during _eos_wait() in basesink,
          the 'eos' flag is overwritten to TRUE after exiting the _eos_wait().
          To resolve the overwritten issue,
          the subclass doing the _eos_wait() call should return the right value.
          If the eos flag is set to TRUE again, it will cause error(enter the eos flow)
          of the following state changing from PAUSED to PLAYING in basesink.
          https://bugzilla.gnome.org/show_bug.cgi?id=754980

2015-10-17 21:47:07 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/playback/gstplaybin2.c:
        * gst/playback/gsturidecodebin.c:
          playbin/uridecodebin: Always post async-done immediately if we're a live pipeline
          Not only if the base class told us, but also if one of our own elements did.
          https://bugzilla.gnome.org/show_bug.cgi?id=756611

2015-10-17 22:25:22 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/playback/gstdecodebin2.c:
        * gst/playback/gstplaybin2.c:
        * gst/playback/gstplaysink.c:
        * gst/playback/gstsubtitleoverlay.c:
          decodebin/playbin/playsink/subtitleoverlay: Post async-done on state change failures
          https://bugzilla.gnome.org/show_bug.cgi?id=756611

2015-10-17 22:20:31 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/playback/gstplaysink.c:
          playsink: Immediately error out if state change fails
          Otherwise we chain up to the parent class' change_state function and might
          override the failure with SUCCESS.
          https://bugzilla.gnome.org/show_bug.cgi?id=756611

2015-10-17 20:36:27 +0300  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/video/video.c:
          video: Add out annotations to the out parameters of gst_video_calculate_display_ratio()
          https://bugzilla.gnome.org/show_bug.cgi?id=754567

2015-10-15 10:01:38 +0900  Vineeth TM <vineeth tm samsung com>

        * gst/playback/gstplaysink.c:
          playsink: Fix volume element leak
          In case sink implements a streamvolume interface, volume element is being got
          from the sink. But this is transfer full. So the memory should be freed before
          setting it to NULL. This was resulting in major memory leaks
          https://bugzilla.gnome.org/show_bug.cgi?id=755867

2015-10-14 00:32:11 +0300  Sebastian Dröge <sebastian centricular com>

        * ext/alsa/gstalsasink.c:
        * ext/alsa/gstalsasrc.c:
          alsa: Use 8 bit pointer type for byte-based pointer arithmetic
          Usually these loops only run once, so there's no problem here. But sometimes
          they run twice, and by adding the number of bytes to a 16 bit pointer type we
          would advance twice as much as we should.
          Also use snd_pcm_frames_to_bytes() in alsasrc to calculate
          the number of bytes to skip, same as we do in alsasink.
          Thanks to Lucio A. Hernandez <lucio a hernandez gmail com> for reporting.

2015-10-04 18:36:00 +0100  Sebastian Dröge <sebastian centricular com>

        * sys/xvimage/xvimagepool.c:
          xvimagesink: Put error message into debug output instead of just throwing it away

2015-09-29 10:12:28 +0530  Rajat Verma <rajat verma st com>

        * gst/playback/gstdecodebin2.c:
          decodebin: free hidden groups at time of switching groups
          hidden groups should be freed at time of switching groups to avoid memory use
          from balloning up.
          https://bugzilla.gnome.org/show_bug.cgi?id=755770

2015-09-30 18:17:13 +0100  Tim-Philipp Müller <tim centricular com>

        * gst/subparse/gstsubparse.c:
          subparse: detect closing tags even if there's a space after the slash
          </ i> should be handled like </i>
          https://bugzilla.gnome.org/show_bug.cgi?id=755875

2015-09-23 11:59:22 -0400  Perry Hung <perry leaflabs com>

        * gst-libs/gst/app/Makefile.am:
          app: pass PKG_CONFIG_PATH for gir files for libgstapp as well
          gir include search directories should respect PKG_CONFIG_PATH,
          just like we do everywhere else. Makes g-i pick up the right
          paths when using ./configure --with-pkg-config-path=
          https://bugzilla.gnome.org/show_bug.cgi?id=755494



Download
========
https://download.gnome.org/sources/gst-plugins-base/1.6/gst-plugins-base-1.6.1.tar.xz (2.70M)
  sha256sum: 9533dcfaa4ee32d435483d9fa88c06b1eba6e9bb234aacd7583f207199f44ba3
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